I'm using the following rtpengine_offer() to force the use of ICE relay and also replace o= and m=
rtpengine_offer("replace-origin replace-session-connection ICE=force-relay RTP");
The SDP is being updated to include an ICE relay candidate, but the IP addresses in the o= and m= lines are not changing to the servers IP address (X.X.X.X).
I also tried using the media-address parameter but that didn't change the behaviour.
When I use rtpengine_answer() then it's not providing an ICE candidate nor changing o= or m=, which I imagine is expected.
What am I missing?
INVITE from the client (before rtpengine)------------------------------------------------------
INVITE sip:server.domain.com SIP/2.0Via: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rportFrom: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>Contact: <sip:username@154.20.1.8:37520;transport=tcp>Call-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Content-Length: 1506Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneSupported: 100rel
--7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp
v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=ptime:20a=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=rtpmap:114 AMR-WB/16000/1a=imageattr:114 octet-align=0a=fmtp:114 octet-align=0a=rtpmap:115 AMR-WB/16000/1a=imageattr:115 octet-align=1a=fmtp:115 octet-align=1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:101 0-16a=pcfg:1 t=1a=sendrecva=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audio
INVITE from the server (after rtpengine)------------------------------------------------------
SIP/2.0 200 OKVia: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rport=37520;received=2.221.73.230From: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>;tag=0ea7681ffd64b0ffcde1bff3393fd505.ae3aCall-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContact: <sip:X.X.X.X:443;transport=tcp>Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 1965
--7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp
v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=imageattr:114 octet-align=0a=imageattr:115 octet-align=1a=pcfg:1 t=1a=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audioa=rtpmap:114 AMR-WB/16000/1a=rtpmap:115 AMR-WB/16000/1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:114 octet-align=0a=fmtp:115 octet-align=1a=fmtp:101 0-16a=sendrecva=ptime:20a=ice-ufrag:kzgbFAnua=ice-pwd:hnEu9EwTrdIZnTWhYbxtmdugvra=candidate:dzkwmPcB28NPHfRA 1 UDP 16777215 X.X.X.X 30366 typ relay raddr X.X.X.X rport 30366a=candidate:dzkwmPcB28NPHfRA 2 UDP 16777214 X.X.X.X 30367 typ relay raddr X.X.X.X rport 30367a=sendrecva=ice-ufrag:uDE9wbvYa=ice-pwd:dx4CcIjqUNFfnRPJEF8GjatXdU
Hi,
In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one.
Could anyone help how to achieve different TCP connections to same destination in kamailio?
thanks,
raj
Hi All,
I don't know if this is the appropriate place to ask the following
questions:
1. Anyone please recommend open-source implementation of PoC
(Push-to-talk Over Cellular) application server? Kamailio can do this?
2. Anyone please recommend open-source implementation of PoC related
media server/replicator?
Thierry Luo
Hi David,
This can solve the issue but depth(records) of that view would be too much.
The architecture is like this:
One kamailio fronting multiple asterisk. Each of asterisk boxes have their
Specific users databases. Thus, I'm
trying to create db connections inside request route or completely write my
own authentication service if I get the user's
Password.
So, that would be great if I would be able to pass the db_url to auth_db
module in request route.
Let me know what you think.
Thanks,
Arish
Hi David,
Thanks for the reply.
I checked out sqlops module, this will solve the problem
only when I get the authentication password (secret)
of the user inside request route. For this I tried checking
kamailio pseudovars but haven't found one. Therefore, I am
sticking with auth_db module.If this is possible by any other way
do let me know.
thanks,
Arish
HI
this is Gaurav
im using kamailio 5.1.2 on ubuntu 18.4
i am trying to configure LetsEncrypt (tls) for kamailio sing following link
https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/
but unable to start kamailio ,i m getting these error
------->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Apr 19 11:23:57 ubuntufresh systemd[1]: Starting Kamailio (OpenSER) - the
Open Source SIP Server...
Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core>
[core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if
you want to use it, load sctp module
Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core>
[core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method
(auto detected)
Apr 19 11:23:58 ubuntufresh kamailio[5365]: Listening on
Apr 19 11:23:58 ubuntufresh kamailio[5365]: Aliases:
Apr 19 11:23:58 ubuntufresh kernel: kamailio[5378]: segfault at 0 ip
00007f8d6fdac646 sp 00007ffe8e2337f8 error 4 in libc-2.27.so
[7f8d6fcfb000+1e7000]
Apr 19 11:23:58 ubuntufresh kamailio[5365]: ERROR: <core>
[core/daemonize.c:303]: daemonize(): Main process exited before writing to
pipe
Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Control process
exited, code=exited status=255
Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Failed with
result 'exit-code'.
Apr 19 11:23:58 ubuntufresh systemd[1]: Failed to start Kamailio (OpenSER)
- the Open Source SIP Server.
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<------------------------------------
plz help
--
*Regards:*
Gaurav Kumar
Following code snippet from default kamailio.cfg never gives 403 if you
smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP.
How to fix it? I want password-based registration (which is OK now) and
permit calls via Kamailio only from permitted IPs.
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself) {
sl_send_reply("403","Not relaying");
exit;
}
Hi All,
I am looking for some assistance on setting up Microsoft Direct Routing
with Kamailio as an SBC. I have a partial config working, I have rewrote
the headers going to Direct routing and can place calls in to and out of
Direct routing. I have problems with Call hang ups and choppy audio only
on the audio stream from Direct Routing. 1) place a call in to direct
routing, Hang the call up from the caller and Direct routing hangs up
the call. But If you place a call in to Direct routing and hang up the
call from direct routing the direct routing hangs up but the call is
still active on the caller. IF you call from direct routing through
kamailio to a callee and hang up from direct routing the call is hung up
correctly. 2) on some calls made to or from direct routing, the audio
stream coming from direct routing is sometimes choppy. Its always the
audio from direct routing never the audio going to direct routing. My
set up: Asterisk <-> Kamailio <-> Direct Routing SIP: Asterisk <->
Kamailio <-> direct routing RTP: Asterisk <-> Direct Routing Kamailio is
rewriting the R-URI, TO, FROM, Contact headers going to Direct routing,
as Microsoft need the sbc hostname to be in the contact header in a
particular format. I am looking for help on the above or if someone
would be kind enough to provide a working config that would be awesome!
:) Many Thanks in advance for assistance. :) Phil
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